Voice encoding/decoding device

ABSTRACT

A voice coding apparatus which can obtain preferable sound quality at a low bit rate is provided. In a mode decision circuit  800  of the voice coding apparatus, a mode is decided from an input voice signal by using a characteristic amount every sub-frame. In a sound source quantization circuit  350 , in case of a predetermined mode, the amplitude or polarity of a non-zero pulse is calculated in advance. Further, combinations of a plurality of shift amounts by which the position of a predetermined pulse is time-shifted and a gain code vector for quantizing a gain are searched. Finally, the combination which minimizes distortion between a reproduced voice and an input voice is selected.

TECHNICAL FIELD

The present invention relates to a voice coding/decoding apparatus forcoding a voice signal at a low bit rate with high quality.

BACKGROUND ART

As a method of efficiently coding a voice signal, for example, a CELP(Code exited linear predictive coding) described in “Code-exited linearprediction: High quality speech at very low bit rates” by M. Schroederand B. Atal (Proc. ICASSP, pp. 937-940, 1985) (Reference 1) is known.Further, “Improved speech quality and efficient vector quantization inSELP” by Klein et al., (Proc. ICASSP, pp. 155-158, 1988) (Reference 2)is known. In these prior arts, on a transmission side, a spectrumparameter representing the spectrum characteristics of a voice signal isextracted from a voice signal every frame (for example, 20 mS) by usinglinear prediction (LPC) analysis. The frame is further divided intosub-frames (for example, 5 mS). Parameters (a delay parametercorresponding to a pitch period and a gain parameter) in an adaptivecode book every sub-frame on the basis of a past sound source signal,and pitch prediction of the voice signal of the sub-frame is performedby using the adaptive code book. For the sound source signal obtained bythe pitch prediction, an appropriate sound source code vector isselected from a sound source code book (vector quantization code book)consisting of noise signals of predetermined types to calculate anappropriate gain, thereby quantizing a sound source signal. Theselection of the sound source code vector is performed such that anerror power between a signal synthesized by a selected noise signal andthe residual signal is minimized. An index representing the type of theselected code vector, a gain, the spectrum parameter, and the parameterof the adaptive code book are combined to each other by a multiplexerunit to be transmitted.

However, in the prior arts described above, an enormous amount ofoperation is required to select an appropriate sound source code vectorfrom the sound source code book. This is because, in the methods ofReferences 1 and 2, a filtering operation or a convolution operation istemporarily performed to code vectors to select a sound source codevector, and the operation is repeated as many times as is equal to thenumber of code vectors stored in the code book. By way of example, it isassumed that the number of bits of the code book is B and that thenumber of dimensions of the code book is N. In this case, when a filteror impulse response length when the filtering operation or theconvolution operation is represented by. K, as an amount of operation,(N·K·2·B·8000)/N is required per second. For example, when B=10, N=40,and K=10, the operation must be repeated 81,920,000 times per second. Asa result, the remarkably enormous amount of operation isdisadvantageously required.

As a method of reducing an amount of operation required to searching asound source code book, for example, ACELP (Algebraic Code Exited LinearPrediction) is proposed. For this method, for example, “16 kbps widebandspeech coding technique based on algebraic CELP” (Proc. ICASSP, pp.13-16, 1991 by C. Laflamme et al., (Reference 3) can be referred to.According to the method of Reference 3, a sound source signal isrepresented by a plurality of pulses, and the positions of the pulsesare represented by the predetermined numbers of bits and transmitted.Here, since the amplitude of each pulse is limited to +1.0 or −1.0, theamount of operation for searching for the pulse can be considerablyreduced. In Reference 3, the amount of operation can be considerablyreduced.

However, although preferable sound quality can be obtained at a bit rateof 8 kB/S or more, when a bit rate lower than the value, and whenbackground noise is superposed on voice, the number of pulses is notsufficient, and the sound quality of a background noise component ofcoded voice is considerably degraded. More specifically, since the soundsource signal is represented by a combination of a plurality of pulses,the pulses are concentrated around a pitch pulse which is a start pointof the pitches in a vowel range of the voice. For this reason, the soundsource signal can be efficiently represented by a small number ofpulses. However, since pulses must be raised at random for a randomsignal such as background noise, it is difficult that the backgroundnoise can be preferably represented by a small number of pulses. Whenthe bit rate is reduced to reduce the number of pulses, sound qualityfor the background noise sharply degraded.

It is, therefore, an object of the present invention to perform voicecoding with a relatively small amount of operation, in particular, smalldegradation of sound quality for background noise even though a low bitrate is set.

DISCLOSURE OF INVENTION

A voice coding apparatus of the present invention includes a vectorquantization circuit for calculating a spectrum parameter of a voicesignal to quantize the spectrum parameter, an adaptive code book circuitfor predicting a voice signal from a sound source signal to calculate aresidual, a sound source quantization circuit for quantizing the soundsource signal by using the spectrum parameter to output the quantizedsound source signal, a gain quantization circuit for quantizing a gainof the sound source signal, a mode decision circuit for extractingcharacteristics from the voice signal to decide a mode, and amultiplexer unit for multiplexing an output from the spectrum parameterquantization circuit, an output from the mode decision circuit, anoutput from the adaptive code book circuit, an output from the soundsource quantization circuit, and an output from the gain quantizationcircuit to output the multiplexed result, wherein, when the output fromthe decision unit represents a predetermined mode, the sound sourcesignal is represented by a combination of a plurality of pulses, theamplitude or polarity of the pulse is calculated from the voice signal,and the sound source quantization unit selects a shift amount and a gaincode vector, which minimize distortion between an input signal and areproduced signal, from combinations of a plurality of shift amounts bywhich the pulses shift and gain code vectors.

The voice decoding apparatus of the present invention also includes ademultiplexer unit for receiving information related to a spectrumparameter, information related to a decision signal, information relatedto an adaptive code book, and information related to a sound sourcesignal to separate the pieces of information from each other, a soundsource signal generation unit for, when the decision signal represents apredetermined mode, generating a sound source signal from an adaptivecode vector, a shift amount of a pulse position, and a gain code vector,and a synthesis filter unit for receiving the sound source signalconstituted by a spectrum parameter to output a reproduced signal. Inthis case, when the decision signal represents a specific mode, pulsepositions may be generated at random, and a sound source signal isgenerated by using the adaptive code vector and the gain code vector.

BRIEF DESCRIPTION OF DRAWINGS

FIG. 1 is a block diagram of a voice coding apparatus according to thepresent invention.

FIG. 2 is an equation expressing distortion generated when linearspectrum pair (LSP) parameter quantization is performed.

FIG. 3 is an equation expressing a response signal x_(z) when an inputsignal is set to be zero (d(n)=0).

FIG. 4 is an equation for calculating a response signal from aperceptual weighting signal.

FIG. 5 is an equation expressing an impulse response of a perceptualweighting filter.

FIG. 6 is an equation for minimizing a delay T corresponding to a pitch.

FIG. 7 is an equation expressing a gain β.

FIG. 8 is an equation for performing pitch prediction.

FIG. 9 is an equation for selecting a combination of a code vector and aposition.

FIG. 10 is an equation for minimizing the equation shown in FIG. 9.

FIG. 11 is another equation for minimizing the equation shown in FIG. 9.

FIG. 12 is a table in which a sound source signal is transmitted suchthat the positions of a plurality of pulses are represented bypredetermined numbers of bits.

FIG. 13 is a table for a specific mode in which a sound source signal istransmitted such that the positions of a plurality of pulses arerepresented by predetermined numbers of bits.

FIG. 14 is an equation showing a polarity for the shift amounts and thepulse positions shown in FIG. 13.

FIG. 15 is an equation for selecting a gain code vector and a shiftamount.

FIG. 16 is an equation for calculating a drive sound source signal.

FIG. 17 is another equation for calculating a drive sound source signal.

FIG. 18 is an equation expressing a response signal.

FIG. 19 is a block diagram of another coding apparatus according to thepresent invention.

FIG. 20 is an equation for selecting a pulse position and a gain codevector.

FIG. 21 is a block diagram of a decoding apparatus according to thepresent invention.

FIG. 22 is a block diagram of another decoding apparatus according tothe present invention.

BEST MODE FOR CARRYING OUT THE INVENTION

The best mode for carrying out the present invention will be describedbelow with reference to the drawings.

(First Embodiment)

FIG. 1 is a block diagram of a voice coding apparatus according to thepresent invention. In FIG. 1, a voice signal is input from an inputterminal 100, and the voice signal is divided by a frame divisioncircuit 110 every frame (for example, 20 mS). In a sub-frame divisioncircuit 120, the voice signal of the frame is divided into sub-frameseach of which is shorter than the frame (for example, 5 mS).

In a spectrum parameter calculation circuit 200, a windows which islonger than a sub-frame length (for example, 24 mS) is applied to thevoice signal of at least one sub-frame to cut a voice, and the spectrumparameter is raised to the power of a predetermined number (for example,P=10th). In the calculation of the spectrum parameter, the known LPCanalysis, a BURG analysis, and the like can be used. In this case, it isassumed that the BURG analysis is used. The details of the Burg analysisare described in “Signal Analysis and System Identification” by Nakamizo(pp. 82 to 87, issued in 1988, Corona Publishing Co., Ltd.) (Reference4) or the like.

In addition, in a spectrum parameter calculation unit, a linearprediction coefficient αil (i=1, . . . , 10) calculated by the Burgmethod is converted into an LSP parameter which is appropriate toquantization or interpolation. Here, with respect to the conversion fromthe linear prediction coefficient into the LSP, “Speech informationcompression by linear spectrum pair (LSP) voice analysis synthesismethod” (Journal of The Institute of Electronics, Information andCommunication Engineers, J64-A, pp. 599-606, 1981) (Reference 5) can bereferred to. For example, linear prediction coefficients calculated bythe BURG method in the second and fourth sub-frames are converted intoLSP parameters, and the LSPs of the first and third sub-frames arecalculated by linear interpolation. The LSPs of the first and thirdsub-frames are subjected to inverse conversion to be returned to linearprediction coefficients, and linear prediction coefficients αil (i=1, .. . , 10, 1=1, . . . , 5) of the first to fourth sub-frames are outputto a perceptual weighting circuit 230. The LSP of the fourth sub-frameis output to a spectrum parameter quantization circuit 210.

In the spectrum parameter quantization circuit 210, the LSP parameter ofa predetermined sub-frame is efficiently quantized, and a quantizationvalue for minimizing distortion expressed by Equation (1) shown in FIG.2.

In this case, LSP (i), QLSP (i) J, and W (i) are an i-th LSP beforequantization, a j-th result after quantization, and a weightingcoefficient, respectively.

In the following description, it is assumed that vector quantization isused as a quantization method and that the LSP parameter of the fourthsub-frame is quantized. As the vector quantization method of an LSPparameter, a known method can be used. As a concrete method, JapanesePatent Application Laid-Open No. 4-171500 (Reference 6), Japanese PatentApplication Laid-Open No. 4-363000 (Reference 7), A Japanese PatentApplication Laid-Open No. 5-6199 (Reference 8), or “LSP Coding UsingVQ-SVQ With Interolation in 4.075 kbps M-LCELP speech coder” by T.Nomura et al., (Proc. Mobile Multimedia Communications, PP. B. 2. 5,1993) (Reference 9) can be referred to.

In the spectrum parameter quantization circuit 210, on the basis of theLSP parameter quantized in the fourth sub-frame, the LSP parameters inthe first to fourth sub-frames are restored. Here, the quantized LSPparameter of the fourth sub-frame of a current frame and the quantizedLSP parameter of the fourth sub-frame of the frame previous to thecurrent frame are linearly interpolated to restore the LSPs of the firstto third sub-frames. In this case, after one type of code vector forminimizing an error power between an LSP before quantization and an LSPafter quantization is selected, the LSPs of the first to fourthsub-frames can be restored by linear interpolation. In order to furtherimprove the performance, after a plurality of code vectors forminimizing the error power are selected as candidates, and accumulateddistortion is evaluated with respect to the candidates, so that acombination of a candidate and an interpolated LSP which minimize theaccumulated distortion can be selected.

The LSPs of the first to third sub-frames restored as described aboveand the quantized LSP of the fourth sub-frame are converted into linearprediction coefficients αil (i=1, . . . , 10, 1=1, . . . , 5) in unitsof sub-frames, and the linear prediction coefficients αil are output toan impulse response calculation circuit 310. An index representing thecode vector of the quantized LSP of the fourth sub-frame is output to amultiplexer 400.

The perceptual weighting circuit 230 receives linear predictioncoefficients αil (i=1, . . . , 10, 1=1, . . . , 5) before quantizationfrom the spectrum parameter calculation circuit 200 in units ofsub-frames, performs perceptual weighting to the voice signals of thesub-frames on the basis of Reference 1, and outputs perceptual weightingsignals.

The response signal calculation circuit 240 receives the linearprediction coefficients αil from the spectrum parameter calculationcircuit 200 in units of sub-frames, and receives the linear predictioncoefficients αil restored by quantization and interpolation from thespectrum parameter quantization circuit 210 in units of sub-frames. Aresponse signal obtained when an input signal is given by zero d (n)=0is calculated for one sub-frame by using a stored value of a filtermemory, and the response signal is output to a subtractor 235. In thiscase, a response signal x_(z) (n) is given by Equation (2), Equation(3), and Equation (4) shown in FIG. 3.

Here, “N” represents a sub-frame length. A reference symbol γ representsa weighting coefficient for controlling an amount of perceptualweighting, and is equal to a value obtained by Equation (7) shown inFIG. 6 to be described later. Reference symbols s w (n) and p (n)represent an output signal from a weighting signal calculation circuitand an output signal of the denominator of a filter of a first term ofthe right-hand side in Equation (7) to be described later, respectively.

The subtractor 235 subtracts a response signal from the perceptualweighting signal for one sub-frame according to Equation (5) shown inFIG. 4, and x′w (n) is output to an adaptive code book circuit 300.

The impulse response calculation circuit 310 calculates an impulseresponse Hw (n) of a perceptual weighting filter in which Z conversionis expressed by Equation (6) shown in FIG. 5 with respect to apredetermined number of points L. Resultant values are output to anadaptive code book circuit 500 and a sound source quantization circuit350.

A mode decision circuit 800 extracts a characteristic amount by using anoutput signal from a frame division circuit, and decides modes in unitsof frames. Here, as characteristics, a pitch prediction gain can beused. Pitch prediction gains calculated in units of sub-frames areaveraged in an entire frame, and the value is compared with a pluralityof predetermined threshold values, so that a plurality of predeterminedmodes are classified. Here, for example, the number of types of modes isset to be 4. In this case, it is assumed that Modes, 0, 1, 2, and 3almost correspond to a silent section, a transition section, a weaklyvoiced section, and a strongly voiced section, respectively. Modedecision information is output to the sound source quantization circuit350, a gain quantization circuit 365, and the multiplexer 400.

In the adaptive code book circuit 500, a past sound source signal v (n),an output signal x′w (n), and a perceptual weighting impulse response Hw(n) are input from the gain quantization circuit 365, the subtractor235, and the impulse response calculation circuit 310, respectively. Adelay T corresponding to a pitch is calculated such that distortionexpressed by Equation (7) shown in FIG. 6 is minimized, and an indexrepresenting the delay is output to the multiplexer 400.

In Equation (8), a reference symbol * represents a convolutionoperation.

A gain β is calculated according to Equation (9) shown in FIG. 7.

In this case, in order to improve the accuracy of delay extraction forfemale voice or child voice, the delay may be calculated as not only aninteger sample, but also a decimal sample value. As a concrete method,for example, “Pitch predictors with high temporal resolution” by P.Kroon et al., (Proc. ICASSP, pp. 661-664, 1990) (Reference 10) can bereferred to. In addition, in the adaptive code book circuit 500, pitchprediction is performed according to Equation (10) shown in FIG. 8, anda prediction residual signal e_(w) (n) is output to the sound sourcequantization circuit 350.

The sound source quantization circuit 350 receives a mode decisioninformation and switches a quantization method for a sound source signaldepending on a mode.

In Modes 1, 2, and 3, it is assumed that M pulses are set. In Modes 1,2, and 3, it is assumed that a B-bit amplitude code book or a polaritycode book for quantizing the amplitudes of the M pulses at once is held.A case in which the polarity code book is used will be described below.The polarity code book is stored in a sound source code book 351.

In a voiced state, the sound source quantization circuit 350 readspolarity code vectors stored in the sound source code book 351,allocates positions to the code vectors, and selects a plurality ofcombinations of code vectors and positions which minimize Equation (11)shown in FIG. 9.

In this equation, a reference symbol Hw (n) represents a perceptualweighting impulse response.

In order to minimize Equation (11) shown in FIG. 9, a combination of apolarity code vector gik and a position mi which minimize Equation (12)shown in FIG. 10 may be calculated.

The combination of the polarity of code vector gik and the position mimay be selected such that Equation (13) shown in FIG. 11 is maximized.This combination further reduces an operation amount required tocalculate the numerator.

In this case, positions at the pulses can be set in Modes 1 to 3 can berestrained as shown in Reference 3. For example, when N=40 and M=5,positions at the pulses can be set are as shown in Table 1 shown in FIG.12.

Upon completion of searching of polarity code vectors, the plurality ofcombinations of polarity code vectors and positions are output to thegain quantization circuit 365.

In a predetermined mode (Mode 0 in this example), as shown in Table 2 inFIG. 13, the positions of the pulses are determined at predeterminedintervals, and a plurality of shift amounts for shifting the positionsof all the pulses are determined in advance. In the following case, fourtypes of shift amounts (Shift 0, Shift 1, Shift 2, and Shift 3) are usedsuch that the positions are shifted by one sample. In this case, theshift amounts are quantized by two bits to be transmitted. In Table 2,shift mount 0 represents the position of a basic pulse. Shift amounts 1,2, and 3 are obtained by shifting the basic pulse position by onesample, two samples, and three samples, respectively. These four typesof shift amounts can be used in this embodiment. However, the types ofshift amounts and the number of shift samples can be arbitrarily set.

Polarities to the shift amounts and the pulse positions of Table 2 shownin FIG. 13 are calculated by Equation (14) shown in FIG. 11 in advance.

The positions shown in Table 2 in FIG. 13 and the polaritiescorresponding thereto are output to the gain, quantization circuit 365in units of shift amounts.

The gain quantization circuit 365 receives mode decision informationfrom the mode decision circuit 800. From the sound source quantizationcircuit 350, a plurality of combinations of polarity code vectors andpulse positions are input in Modes 1 to 3, and combinations of pulsepositions and polarities corresponding thereto are input in units ofshift amounts in Mode 0.

The gain quantization circuit 365 reads a gain code vector from a gaincode book 380. In Modes 1 to 3, the gain quantization circuit 365searches the selected plurality of combinations of polarity code vectorsand position for a gain code vector such that Equation (15) shown inFIG. 14 is minimized. A gain code vector for minimizing distortion andone type of combination of a polarity code vector and a position areselected.

Here, a case in which both the gain of an adaptive code book and thegain of a sound source represented by pulses are simultaneouslyvector-quantized is exemplified. An index representing the selectedpolarity code vector, a code representing a position, and an indexrepresenting a gain code vector are output to the multiplexer 400.

When the decision information is Mode 0, a plurality of shift amountsand polarities corresponding to the positions in the respective shiftamounts are input to search for a gain code vector, and a gain codevector and one type of shift amount are selected such that Equation (16)shown in FIG. 15 is minimized.

Here, reference symbols βk and G′k represents the Kth code vector in atwo-dimensional gain code book stored in the gain code book 380.Reference symbol δ(j) represents the j-th shift amount, and thereference symbol g′k represents the selected gain code vector. An indexrepresenting the selected code vector and a code representing a shiftamount are output to the multiplexer 400.

In Modes 1-3, a code book for quantizing the amplitudes of a pluralityof pulses can be trained in advance by using a voice signal to bestored. As the method of learning a code book, for example, “AnAlgorithm for vector quantization design” by Linde rt al., (IEEE Trans.Commun., pp. 84-95, January, 1980) (Reference 11) can be referred to.

The weighting signal calculation circuit 360 receives mode decisioninformation and indexes, and reads code vectors corresponding theindexes from the indexes. In Modes 1 to 3, a drive sound source signal V(N) is calculated on the basis of Equation (17) shown in FIG. 16.

The signal v (n) is output to the adaptive code book circuit 500.

In Mode 0, a drive sound source signal v (n) is calculated on the basisof Equation (18) shown in FIG. 17.

The signal v (n) is output to the adaptive code book circuit 500.

Response signals s_(w) (n) are calculated for sub-frames by Equation(19) shown in FIG. 18 by using an output parameter from the spectrumparameter calculation circuit 200 and an output parameter from thespectrum parameter quantization circuit 210, and are output to theresponse signal calculation circuit 240.

(Second Embodiment)

FIG. 19 is a block diagram of another coding apparatus according to thepresent invention. Since constituent elements in FIG. 19 to which thesame reference numerals as in FIG. 1 are added perform the sameoperations as in FIG. 1, a description thereof will be omitted. In FIG.19, the operation of a sound source quantization circuit 355 isdifferent from that of FIG. 1. In this case, when mode decisioninformation is Mode 0, a position generated according to a predeterminedrule is used as a position of a pulse.

For example, the positions of pulses the number of which arepredetermined (for example, M1) are generated by a random numbergeneration circuit 600. More specifically, M1 numeral values generatedby the random number generator are considered as the positions ofpulses. In addition, the plural sets of positions of different types aregenerated. The M1 positions of the plural sets generated as describedabove are output to the sound source quantization circuit 355.

When the mode decision information is Modes 1 to 3, the sound sourcequantization circuit 355 performs the same operation as that of thesound source quantization circuit 350 shown in FIG. 1. In Mode 0,polarities are calculated from Equation (14) in advance for the pluralsets of positions output from the random number generation circuit 600.

The plural sets of positions and the polarities corresponding to pulsepositions are output to a gain quantization circuit 370.

The gain quantization circuit 370 receives the plural sets of positionsand the polarities corresponding to the pulse positions, searches for acombination of gain code vectors stored in the gain code book 380, andselects one type of combination of a set of positions and a set of gaincode vectors which minimize Equation (20) shown in FIG. 20 to output thecombination.

(Third Embodiment)

FIG. 21 is a block diagram of a decoding apparatus according to thepresent invention. This decoding apparatus may be combined to the codingapparatus shown in FIG. 1 to form a coding/decoding apparatus. In FIG.21, a demultiplexer 500 receives mode decision information, an indexrepresenting a gain code vector, an index representing delay of anadaptive code book, information of a sound source signal, an index of asound source code vector, and an index of a spectrum parameter from areceived signal, and separately outputs the respective parameters.

A gain decoding circuit 510 receives the index of the gain code vectorand the mode decision information, and reads and outputs a gain codevector from the gain code book 380 depending on the index.

An adaptive code book circuit 520 receives the mode decision informationand the delay of the adaptive code book, generates an adaptive codevector, and multiples the gain code vector by the gain of the adaptivecode book to output the resultant value.

In a sound source signal restoration circuit 540, when the mode decisioninformation is Modes 1 to 3, a sound source signal is generated by usinga polarity code vector read from a sound source code book 351,positional information of pulses, and the gain code vector to output thesound source signal to an adder 550.

When the mode decision information is Mode 0, the sound source signalrestoration circuit 540 generates a sound source signal from a pulseposition, a shift amount of the position, and the gain code vector tooutput the sound source signal to the adder 550.

The adder 550 generates a drive sound source signal V (N) by using anoutput from the adaptive code book circuit 520 and an output from thesound source signal restoration circuit 540 on the basis of Equation(17) in Modes 1 to 3 or on the basis of Equation (18) in Mode 0 tooutput the drive sound source signal v (n) to the adaptive code bookcircuit 520 and a synthesis filter circuit 560.

A spectrum parameter decoding circuit 570 decodes a spectrum parameterto convert the spectrum parameter into a linear prediction coefficient,and outputs the linear prediction coefficient to the synthesis filtercircuit 560.

The synthesis filter circuit 560 receives the drive sound source signalv (n) and the linear prediction coefficient, calculates a reproducedsignal, and outputs the reproduced signal from a terminal 580.

(Fourth Embodiment)

FIG. 22 is a block diagram of another decoding apparatus according tothe present invention. This decoding apparatus may be combined to thecoding apparatus shown in FIG. 2 to form a coding/decoding apparatus.Since constituent elements in FIG. 22 to which the same referencenumerals as in FIG. 21 perform the same operations as in FIG. 21 areadded perform the same operations as in FIG. 21, a description thereofwill be omitted.

In FIG. 22, when mode decision information is Modes 1 to 3, a soundsource signal restoration circuit 590 generates a sound source signal byusing a polarity code vector read from a sound source code book 351,positional information of pulses, and a gain code vector to output thesound source signal to the adder 550. When the mode decision informationis mode 0, the positions of pulses are generated from the random numbergeneration circuit 600, and a sound source signal is generated by usingthe gain code vector to output the sound source signal to the adder 550.

INDUSTRIAL APPLICABILITY

According to the present invention described above, in a predeterminedmode, the number of pulses can be considerably increased in comparisonwith a conventional method. For this reason, even though voice on whichbackground noise is superposed is coded at a low bit rate, a backgroundnoise component can be preferably coded and decoded.

1. A voice coding apparatus comprising: a spectrum quantization circuitfor calculating and quantizing a spectrum parameter of a voice signal;an adaptive code book circuit for predicting said voice signal from asound source signal to calculate a residual; a sound source quantizationcircuit for quantizing said sound source signal by using said spectrumparameter to output the quantized sound source signal; a gainquantization circuit for quantizing a gain of said sound source signal;a mode decision circuit for extracting characteristics from said voicesignal to decide a mode; and a multiplexer unit for multiplexing anoutput from said spectrum parameter quantization circuit, an output fromsaid mode decision circuit, an output from said adaptive code bookcircuit, an output from said sound source quantization circuit, and anoutput from said gain quantization circuit to output the multiplexedresult, characterized in that: when the output from said mode decisioncircuit represents a predetermined mode, said sound source signal isrepresented by a combination of a plurality of pulses wherein anamplitude or polarity of the pulse is calculated from said voice signal;and said sound source quantization unit selects a shift amount and acode vector, which minimize distortion between an input signal and areproduced signal, from combinations of a plurality of shift amounts bywhich the pulses shift and gain code vectors.
 2. The voice codingapparatus according to claim 1, characterized in that the positions ofthe pulses the number of which is predetermined are arranged atpredetermined intervals, and a plurality of shift amounts for shiftingthe positions of the pulses as a whole are determined.
 3. The voicecoding apparatus according to claim 1, characterized in that thecombinations of the positions of the pulses the number of which ispredetermined are generated at random, and the plurality of combinationsare determined.
 4. A voice decoding apparatus characterized bycomprising: a demultiplexer unit which receives information related to aspectrum parameter, information related to a decision signal,information related to an adaptive code book, and information related toa sound source signal to separate the pieces of information from eachother; a sound source signal generation unit adapted to generate a soundsource signal from an adaptive code vector, a shift amount of a pulseposition, and a gain code vector when the decision signal represents apredetermined mode; and a synthesis filter unit which receives the soundsource signal constituted by a spectrum parameter to output a reproducedsignal.
 5. A voice decoding apparatus characterized by comprising: ademultiplexer unit which receives information related to a spectrumparameter, information related to a decision signal, information relatedto an adaptive code book, and information related to a sound sourcesignal to separate the pieces of information from each other; a soundsource signal generation unit adapted to generate positions of pulsesrepresenting sound source signals at random and generating a soundsource signal by using an adaptive code vector and a gain code vectorwhen the decision signal represents a specific mode; and a synthesisfilter unit which receives the sound source signal constituted by aspectrum parameter to output a reproduced signal.